package com.cloudwebrtc.voip.sipenginev2;

import com.cloudwebrtc.voip.mediaengine.MediaStream;

public interface Call
{
	public void Accept();

	public void Accept(boolean send_audio,boolean send_video);
	
	public void Reject(int code, String reason);

    public int Hangup();

    public int UpdateCall(boolean  enable_video);

    public int Hold();

    public int UnHold();
    
    public int SendDtmf(DtmfMethod dtmf_method, String tone, boolean play_dtmf_tone);

    public String GetCallerId();
	
	public Direction GetDirection() ;
	
	public CallState GetCallState();
	
	public boolean GetSupportVideo();
	
	public boolean GetSupportData();
	
	public SipProfile GetProfile();
	
	public MediaStream GetMediaStream();
	
    public int GetErrorCode();

    public String GetErrorReason();
    
    public  CallReport GetCallReport();
    
    public String CallStateName(CallState state);
    
    public String GetUniqueId();
    
    public String GetDeviceType();
}
